For many years voice telephone service was implemented over a circuit switched network commonly known as the public switched telephone network (PSTN) and controlled by a local telephone service provider. In such systems, the analog electrical signals representing the conversation are transmitted between the two telephone handsets on a dedicated twisted-pair-copper-wire circuit. More specifically, each telephone handset is coupled to a local switching station on a dedicated pair of copper wires known as a subscriber loop. When a telephone call is placed, the circuit is completed by dynamically coupling each subscriber loop to a dedicated pair of copper wires between the two switching stations.
More recently, the copper wires, or trunk lines between switching stations have been replaced with fiber optic cables. A computing device digitizes the analog signals and formats the digitized data into frames such that multiple conversations can be transmitted simultaneously on the same fiber. At the receiving end, a computing device reforms the analog signals for transmission on copper wires. Twisted pair copper wires of the subscriber loop are still used to couple the telephone handset to the local switching station.
More recently yet, voice telephone service has been implemented over the Internet. Advances in the speed of Internet data transmissions and Internet bandwidth have made it possible for telephone conversations to be communicated using the Internet's packet switched architecture and the TCP/IP protocol.
Software is available for use on personal computers which enable the two-way transfer of real-time voice information via an Internet data link between two personal computers (each of which is referred to as an end point or client). Each end point computer includes appropriate hardware for driving a microphone and a speaker. Each end point operates simultaneously both as a sender of real time voice data and as a receiver of real time voice data to support a full duplex voice conversation. As a sender of real time voice data, the end point computer converts voice signals from analog format, as detected by the microphone, to digital format. The software then facilitates data compression down to a rate compatible with the end point computer's data connection to an Internet Service Provider (ISP) and facilitates encapsulation of the digitized and compressed voice data into a frame compatible with the user datagram protocol and internet protocol (UDP/IP) to enable communication to the other end point via the Internet.
As a receiver of real time voice data, the end point computer and software reverse the process to recover the analog voice information for presentation to the operator via the speaker associated with the receiving computer.
To promote the wide spread use of Internet telephony, the International Telephony Union (ITU) had developed a set of standards for Internet telephony. The ITU Q.931 standard relates to call signaling and set up, the ITU H.245 standard provides for negotiation of channel usage and compression capabilities between the two endpoints, and the ITU H.323 standard provides for real time voice data between the two end points to occur utilizing UDP/IP to deliver the real time voice data.
Additionally, the Internet Engineering Task Force (IETF) has developed a set of standards for initiating real time media data sessions known as the Session Initiation Protocol (SIP). SIP provides for UDP/IP messages to be exchanged between the two endpoints (or between the two endpoints and multiple proxy servers) to provide for call signaling and negotiation of compression capabilities.
A problem associated with standard ITU Internet telephony and with SIP Internet telephony is that network address translation (NAT) firewalls prevent the transmission of UDP/IP frames from an endpoint computer outside the firewall to an endpoint computer on a private network inside the firewall.
More specifically, both the ITU Internet telephony standards and the SIP standards provide for each endpoint to designate a real time transport protocol (RTP) channel, which comprises an IP address and port number, for receipt of media datagrams and to provide that RTP channel to the other end point.
Because the private network client does not have a globally unique IP address, a frame sent to such non-globally unique IP address can not be routed on the Internet and will be lost. Further, even if the private network client were able to identify and designate the IP address of the NAT firewall, the private network client has no means for establishing a port on the NAT firewall for receipt of media datagrams.
Because of the wide spread use of NAT firewalls which typically provide both IP address translation and port translation of all frames sent from the private network to the Internet, what is needed is a system and method for establishing and maintaining Internet telephony conversations between two clients, both of which are located on private networks behind NAT firewalls. What is also needed is a system and method for establishing and maintaining Internet telephone conversations between a client located on a private network behind a NAT firewall and a client with an Internet routable IP address (e.g. public IP address on the Internet) that operates a receiving UDP channel that is different from its sending UDP channel.